The following is at least applicable to Mac OS X Mountain Lion 10.8.5 in 2013.

Rule 1: there is no point having "high definition" audio resources if you don't
have your playback device set to handle high definition audio or if you don't
play them back on very good quality speakers or very good quality headphones.

What's the point of all those endless arguments about "best" and "minimal acceptable" sample rates and bit depths if your system is not even setup right to hear the difference ? It seems most Macs since many years before 2013 have supported at least 2-channel 24bit Integer sample bit depth at 96kHz sample rate, which is certainly higher definition than sleepy old stereo 16-bit 44.1kHz "CD quality", even if there is a lot of empirical evidence and psycho-acoustic science that proves you may never be able to notice the difference. So I give here some tips and links for checking "high definition" audio settings on your Mac.

But firstly, one of the curious things about the rough term "high definition audio" is that it is, well, not so well defined. You will often find it on the web loosely to mean 2-channel 24-bit 96kHz, while sometimes it is used for as low as 2-channel 24-bit 48kHz or even 44.1kHz, as long as 24bits per sample (per channel) are used. Some extreme audiophiles insist it is not true high definition unless it is at least 2-channel 24-bit 192kHz (and they probably also listen to ultrasonic "whistle music" at home together with their pet dogs and cats).

'Intel High Definition Audio (also called HD Audio or Azalia) refers to the specification released by Intel in 2004 for delivering high-definition audio ..

Hardware based on Intel HD Audio specifications is capable of delivering 192-kHz 32-bit quality for two channels, and 96-kHz 32-bit for up to eight channels. However, as of 2008, most audio hardware manufacturers do not implement the full high-end specification, especially 32-bit sampling resolution.

.. Mac OS X has full support with its AppleHDA driver. ..'

Well I'm not sure my beloved old MacBook Pro, Early 2008 17" can handle all that, but it certainly offers up to 2-channel 24-bit Integer 96kHz on built-in Microphone, built-in Input, and Built-in Output. You can check this for your system using a special application Utilities > Audio Midi Setup:

Whatever sample rate part of "Format" you choose there will also (on reload of the system info window) be reflected under About this Mac > System Report ...> Hardware > Audio > Devices, but it says nothing there about the sample bit depth:

Here you see I have now increased the sample rate for the output (only) to the maximum available for my system, 96kHz. The System Report is however strangely lacking in detail on the Intel High Definition Audio section (as is Apple's spec page for my Mac Book Pro 2008):

I have so far been able to find barely anything else concrete online about the audio hardware of any of my Mac machines, and in particular, I can't find anything about the credentials of the input ADCs - one good reason, especially when recording live music tracks, to instead use an external audio interface, say over USB or FireWire, with well known specifications instead, see discussion below.

The Blu-Ray and HD DVD formats are capable of up to 48Mbps. Around 30Mbps of this transfer speed is reserved for video, leaving a sizeable chunk for (uncompressed) audio.

These audio streams can be sent to an AV receiver/amplifier as bitstream (encoded digital data) or PCM (essentially raw digital data.) Bitstreamed audio from a DVD, Blu-Ray or HD DVD disc needs to be decoded. This can sometimes be done by the player and output as PCM to the amplifier/receiver. More often than not though, the decoding is done by an AV receiver/processor. Regardless of which method you use, there is no difference in quality between PCM and lossless bitstreamed formats like Dolby True HD and DTS HD Master Audio. As a result, many Blu-Ray and HD DVD discs will offer both Dolby True HD/DTS HD Master Audio and (multi-channel) PCM soundtracks for the sheer convenience.

Along with the lossless Dolby True HD and DTS HD Master Audio formats, Blu-Ray and HD DVD offer Dolby Digital Plus and DTS HD High Resolution. While being a “lossy” format, these other two new standards offer benefits that Dolby Digital and DTS from DVD discs can’t such as higher sample rates.'

For audio, BD-ROM players are required to support Dolby Digital (AC-3), DTS, and linear PCM. Players may optionally support Dolby Digital Plus and DTS-HD High Resolution Audio as well as lossless formats Dolby TrueHD and DTS-HD Master Audio. BD-ROM titles must use one of the mandatory schemes for the primary soundtrack. A secondary audiotrack, if present, may use any of the mandatory or optional codecs.

Phew, well that made it easier. Basically, all Blu-ray players can at least support:

In general, the higher sample frequencies are only available when no more than 6 channels are used.

So every BD-ROM player is required to at least support 24-bit at 192kHz (on 6 channels), which is higher than the 24-bit at 96kHz on 2 channels my old 2008 MacBook Pro can handle. It is however not as high as the 32-bit requirement for Intel High Definition Audio (see top).

Sourcing high definition audio (free)

Ok, now we know what high definition audio is (roughly) and how to ensure the Mac system audio is set for its audio highest definition capability, let's get some into our Mac. There are at least the following ways:

1. Record some yourself. Definitely the most fun and instructive, and the main subject for rest of this article. This way, as long as you do it right, because you get to explore the noise level, you know it is not only a high definition recording it is also a high definition source.

Especially useful are the generated ones, since you know there is not a lot of noise in them (or you know what kind of noise is in them).

Amongst the ones based on live recorded music, I found the free Steinway and Sons piano ones particularly interesting (as I am a piano player) and I confess I found it very hard in quality headphones to hear any difference between the CD quality 16-bit 44.1kHz WAV files and the 24-bit 96kHz WAV files.

3. Download free legal examples of high definition audio from all over the web (not necessarily professionally prepared audio test files, just any music). There is a good discussion of high definition audio resources here: How to find and play high-resolutions audio on the Mac, Jun 2011, by Kirk McElhearn. Includes also an excellent description of the inclusion of high resolution audio for sale on iTunes and other sites, and some of the lossless compression formats like FLAC (FLAC-HD) and ALAC often used to distribute them:

'Playing high-res files

Macs can natively support up to 24/96, played through iTunes or other software. However, without a couple settings tweaks, audio files with resolution higher than 16-bit/44.1kHz will automatically be downsampled to that resolution. So the first thing you need to do is set your sound output to 24/96. To do so, open Audio MIDI Setup, found in /Applications/Utilities. Select the desired output on the left, and then change the settings in the Format section on the right to 96000.0 kHz and 2ch-24bit.

Once you’ve made this change, you can play files at any resolution up to and including 24/96; lower-resolution files will actually be upsampled to 24/96 (which, unfortunately, won’t make them sound any better.)'

But you will never know for sure whether the actual music/sound source was in fact "high quality" and worth the high resolution audio treatment. The only way to be sure of that is to either record it yourself (very carefully), or get professional audio test files.

4. Steal illegal high resolution audio resources from all over the web (like high definition FLAC popular on torrent sites). I'll ignore this one. Besides, you can't be sure of what you get anyway !

5. Rip illegal high resolution audio resources from say Blu-rays at home. I'll ignore this one too. And you usually still can't be sure what the specs of the source behind the mastered audio put on the Blu-ray were, or the quality of the source, even if a "pro" did it for a major production house and a major entertainment distributor.

CAUTION: just because music samples are distributed using a high definition audio format does not mean the music is in fact "high definition audio". It is in some cases no more than a cynical marketing exercise. This is also sometimes true of so-called "high definition samples" offered to unsuspecting computer music composers who then work in a higher definition mode like 32-bit floating point (nevertheless of benefit from the point of view of internal processing, mixing, and FX application) and 96kHz sample rate in their DAW, but are in fact still working with music of no better than CD quality 16-bit 44.1kHz !

Professionally prepared audio test files are best for exploring high definition audio !

But it's fun trying to create your own, and one can learn a lot by doing it, so let's focus for the rest of this article on creating your own high resolution audio recordings.

Recording high(er) definition audio from live sources

Obviously, if the sound/music source is not clean with low ambient noise and your equipment is not clean with good specifications, there is little point. I found it however instructive to experiment with it even if there is some source or equipment noise clearly present to explore how the higher resolution recordings handle it.

Some remarks on audio interfaces

If you are recording it's no use having 24-bit depth at 96kHz on 2 channels if you can't get music into the machine at that quality. Many Mac model audio ports support both (through different connectors) analog via RCA and digital S/PDIF via TOSLINK with round mini-adapter (which is almost RCA shaped). From Wikipedia: TOSLINK:

'Also known generically as an "optical audio cable" or just "optical cable", its most common use is in consumer audio equipment (via a "digital optical" socket), where it carries a digital audio stream from components such as CD and DVD players, DAT recorders, computers, and modern video game consoles, to an AV receiver that can decode two channels of uncompressed lossless PCM audio or compressed 5.1/7.1 surround sound such as Dolby Digital Plus or DTS-HD High Resolution Audio. Unlike HDMI, TOSLINK does not have the capacity to carry the lossless versions of Dolby TrueHD and DTS-HD Master Audio.'

'S/PDIF can carry two channels of uncompressed PCM audio or compressed 5.1/7.1 surround sound (such as DTS audio codec) with a maximum bandwidth of 3.072 Mbit/s per channel for a total of 6144 kbit/s; it cannot support uncompressed lossless formats (such as Dolby TrueHD and DTS-HD Master Audio) which require greater bandwidth like that available with HDMI or DisplayPort.'

But what's the use of Mac audio ports for recording from analog signals if Apple don't publish the A/D specs of specific machines ?

External audio interfaces

Another way of getting high quality audio sources for recording in - with known specs - is through an external audio interface to USB2/3, or FireWire400/800 (aka IEEE 1394). My MacBook Pro early 2008 has 3 USB2 ports and:

One FireWire 400 port at up to 400 Mbps

One FireWire 800 port at up to 800 Mbps

There have been lots of arguments online about whether PC cards are better than external interfaces, but with stable modern USB2/3 or FireWire and decent cables you will be fine (and for most Macs there is no choice, external it is). And there are lots of other nice things you can do with external audio interfaces. For example, some of them have very high quality mic preamps and stable phantom power. Some also have nice rerouting and inline FX capabilities.

I have an old M-Audio Firewire 410 (from about 2006), which also has excellent GUI software support, but the specs look a bit tired compared with modern interfaces:

M-Audio FireWire 410

• Dual low-noise mic/instrument preamps with gain controls, LED metering, phantom power and 66dB of available gain

BTW quoting SNR in negative (-)dB for such equipment is wrong according to this guide from RANE on audio specifications, and is often confused with EIN. Equivalent Input Noise or Input Referred Noise, which can be specified as, for example: EIN = -130 dBu, 22 kHz BW, max gain, Rs = 150 ohms.]

However compare 108dB SNR with some more modern audio interfaces/cards like the M-Audio Delta Audiophile 192 with an input SNR of 113dB, or the ASUS Xonar Essence STX with Input SNR of 118dB, and my old M-Audio Firewire 410 interface certainly seems well out of date. The modern cards and interfaces typically also have more generous frequency ranges from 10Hz to 90kHz (presumably for recording very big church organs and coyote howls).

Borrowing the excellent diagram from ZedBee's super article Digital recording rule of thumb, you can see that as long as you record (if using the 24-bit EBU Digital standard) with the RMS around -18dBFS, even 108dB SNR is not too bad (certainly good enough for even "high definition" home recording projects):

There is an excellent guide to Choosing A PC Audio Interface by Martin Walker from Sound on Sound mag from Nov 2004. The basic points are still relevant, with one of the major rules broken by my older M-Audio Firewire 410 (namely it uses only -10dBV consumer level voltage, not +4dBu pro level):

"Consumer & Professional

Many musicians are still confused about which interface input sensitivity and output level to use when faced with choices of [-]10dBV (consumer) or +4dBu (professional). It's easy to get bogged down in discussing millivolts and so on, but there are a few simple rules of thumb that should make everything easier to understand.
Always stick to the '+4' option if you can, since this generally results in lower noise levels. If you can't get high enough recording levels with '+4' input sensitivity on your interface, and there's no -10/+4 switch on the source gear, switch to '-10'. Similarly, stick with +4 output levels unless any connected gear can't cope with these higher levels, in which case revert to '-10'."

Aside: note carefully that these are in different dB scales, +4dBu and -10dBV (although product specs often state just +4dB or -10dB). From Understanding DB:

+4dBu equals 1.23 Volts RMS. Actually 1.2276 V

The reference level of -10dBV (0.316 V) is the equivalent to a level of -7.8dBu.

+4dBu and -10dBV systems have a level difference of 11.8 dB and not 14 dB. This is almost a voltage ratio of 4:1

Martin Walker seems to agree that one doesn't need the world's best signal-to-noise and dynamic range to make decent high quality recordings (although the needs of a true audio pro are more demanding):

'The most hotly quoted specification for any audio interface tends to be its dynamic range or signal/noise ratio. There's still a lot of confusion about these two terms, and this is hardly surprising considering each may be measured in a variety of ways. However, the way audio interface manufacturers measure them seems to be reasonably consistent, and using these particular methods the two figures also tend to be very similar with many products, which makes products that quote one or the other easier to compare.

In audio interface terms, Signal/Noise ratio compares the maximum signal level that you can send to the interface (ie. that which makes the input meters just register 0dB) with the background noise level when no signal is present. However, some crafty soundcard manufacturers realised early on that they could achieve amazingly good s/n figures by automatically muting the output in the absence of an input signal, so that its background noise level was significantly lower. The audio interface dynamic range measurement therefore measures the background noise level in the permanent presence of a low-level signal (generally a 1kHz sine wave at -60dBFS), which is subsequently notched out using a filter. Dynamic range is therefore a slightly more reliable real-world test. You may spot some cheap soundcards with significantly worse results for their dynamic range than for their Signal/Noise (S/N) ratio.

..

Both figures are generally measured via an 'A'-weighting network, which rolls off the noise either side of its 3kHz centre frequency, in line with the sensitivity of the human ear. In essence, a 'dBA' rating reflects more closely how annoying we will find the background noise, with low-level hums below 200Hz and whistles above 10kHz being less obvious than hiss between about 1kHz and 6kHz. A dBA rating is generally a few dBs better than a 'flat' measurement.

Despite the fact that most audio recordings still end up on a Red Book Audio CD at 16-bit/44.1kHz, most of us have abandoned 16-bit recording and playback in favour of the wider dynamic range possible with 24 bits. A typical soundcard will provide a maximum dynamic range of 96dBA at 16-bit, but well over 100dBA when using 24-bit, which allows us to worry less about taking our recordings to within a few dB of clipping, because the background noise levels are so much lower.

However, when comparing the dynamic ranges of different audio interfaces, don't lose sight of the signals you'll be recording. If, like me, you still record the outputs of various hardware synths, the chances are that they won't have a dynamic range of more than about 80dB. If you're capturing a live performance via a mic, the background noise level of that mic and its associated preamp may already be higher than that of the audio interface, especially since it's difficult to make recording areas really quiet without extensive soundproofing. After all, as Hugh Robjohns said in SOS September 2004: "In most public venues I find the ambient noise floor is typically about 50-55dB below the peak level of a modest orchestra, organ, or choral group".

So, while buying an interface with the lowest possible background noise is sensible, in the real world many musicians won't be able to hear any difference at normal listening levels between interfaces with a dynamic range of 110dBA and 120dBA. Moreover, I've recently spotted various musicians grumbling about the background noise levels of specific soundcards, when they were actually hearing digital nasties due to the the effects of a ground loop. As soon as they modified their wiring or introduced a DI box to deal with the problem, most were amazed at how quiet a background noise level of 100dBA was!'

Yep, that last one happened to me once, too. Buzz buzz, and it was it just a bad (dedicated) mic preamp with a ground loop. And Martin Walker takes away some more worry:

'It's also worth pointing out that switching to 32-bit recording and playback in your audio application won't result in an even larger dynamic range — the benefit of the 32-bit float format is massive internal headroom and no possibility of internal clipping when mixing together loads of tracks, but the interface will still have 24-bit converters on the input and output. Unless the world suddenly becomes a much quieter place, 24 bits will remain quite sufficient to digitise it.'

I found the following relevant, because I have an old 1993 Roland RD500 digital piano/organ with synth sounds (although I could not find out any specs regarding noise, or about how it produces its sounds or equivalent sample bit depths if it uses waveform reconstruction):

'Sample Rate Wars

While even budget audio interfaces are now beginning to feature 192kHz sample rates, there are still arguments raging on most audio forums about whether or not it's worth moving from a sample rate of 44.1kHz to 48, 88.2 or 96kHz. Many musicians stick to 24-bit/44.1kHz because they still create their music largely with hardware MIDI synths and soft samplers that themselves use 44.1kHz samples, so they see little point in moving higher, especially as they intend the final mix to end up on a 16-bit/44.1kHz audio CD. However, even those using electronic sources will probably find subsequent compression and peak limiting more accurate at higher sample rates, while EQ tends to sound far more analogue in nature and metering is more accurate. Those using soft synths that calculate or otherwise model their waveforms may also find they sound cleaner.

For live classical and other acoustic recordings I suspect most serious engineers now prefer 24-bit/96kHz, particularly if the final recordings are for DVD release at 48 or 96kHz ..

.. mainstream PC magazines may mark a particular review soundcard down if it doesn't offer a 192kHz sample rate, I personally consider this option a huge red herring in the case of most audio interfaces under £500. If you can hear the improvement, use 192kHz, but bear in mind that the rest of the signal chain needs to be of extremely high quality to really exhibit any benefit over 96kHz.

Remember, also, when choosing a sample rate for your projects, that at 192kHz every plug-in and soft synth you run will consume over four times as much CPU overhead, occupy more than four times the amount of hard disk space, and cut your potential simultaneous track count by more than a factor of four over 44.1kHz.'